Forget it, it works now !! Had to add "pridialplan=unknown" to make it work ...
Thanks anyway :-) On Fri, 2003-10-10 at 22:24, Marcel Prisi wrote: > Hi ! > > I'm not sure if it's the right list but I hope someone can help me ! > > I am setting up a SIP VoIP gateway using Asterisk and a Digium E100P > card plugged on a Swisscom PRI ISDN. > > I setup my card using ccs framing, hdb3 coding with crc4 checking and > euroisdn switch type. > > Everything works well with incoming calls, my Grandstream SIP phones > ring and answer calls, show right CID and so on, but when dialling out > from SIP to PSTN, it looks like it works, asterisk shows the correct > things, says number is ringing, but nothing happens on the remote PSTN > phone. > > Asterisk has no "tone zone" for Switzerland, means no DTMF info. I tried > several other countries DTMF settings with no success thinking it might > be the cause. > > Any suggestion ? Do you know Swisscom's DTMF settings ? Similar to other > countries ?? > > Thanks. > > > > ---------------------------------------------- > [EMAIL PROTECTED] Maillist-Archive: > http://www.mail-archive.com/swinog%40swinog.ch/ -- :: Marcel Prisi / Technical Manager ------------------- - - - - - - - virtua.ch web solutions Ruelle du Soleil levant 6 CH - 1170 Aubonne T. +41 21 807 28 00 F. +41 21 807 28 01 ---------------------------------------------- [EMAIL PROTECTED] Maillist-Archive: http://www.mail-archive.com/swinog%40swinog.ch/
